Hi everyone,
I have some issues about Asterisk RTP traffic. We have the following call-routing topology:
SBC → Asterisk 1 (Work as a loadbalancer, route call to other asterisk) → Asterisk IVR (Our ivr runs via AGI We have 4 seperate servers) → Asterisk 1 → Genesys Santral
Asterisk 1 distributes incoming calls across our IVR servers. The IVR servers run AGI applications and when agent required, route the call back through the Asterisk 1 toward genesys.
We are experiencing RTP packet loss that appears to be associated with the Asterisk 1. We would like to remove this server from the media path while retaining it in the SIP signalling path, if possible.
Our desired media paths are:
During IVR: SBC ↔ Asterisk IVR node
After transfer: SBC ↔ Genesys-side media endpoint
Asterisk version: 22.1.1
Channel: PJSIP
Current IVR-to-Genesys handoff: Dial/PJSIP
Current direct media configurations;
The trunk to the SBC on Asterisk1
[SBC_IST_154]
type=endpoint
transport=transport-udp5062
disallow=all
allow=ulaw,alaw
aors=sbcist154_aors
direct_media=no
context=saha_inbound… following aors
The trunk to the Asterisk IVRs on Asterisk 1
[ast188]
type=endpoint
transport=transport-udp5062
disallow=all
allow=alaw,ulaw
direct_media=no
aors=ast188-aors
context=transfer
Our preference is to solve this within the existing Asterisk topology. If that is not reliable for this call flow, we are also considering moving SIP load balancing to Kamailio.
Thanks for any guidance, especially from anyone running a similar SBC–Asterisk IVR–Genesys integration.